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Help to configure SIP Trunk in CUCM 9.1

Unanswered Question
Oct 8th, 2013
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Hi everyone.


I would like someone could help me to configure a SIP Trunk.


I have to connect a CUCM 9.1 to a  VoIP service provider. This SP have send me the next informatio.


a Host name of the SIP Trunk


a User and Password.


I have nerver configured a sip trunk with authentication in CUCM so I have investigated about it and I'd tried to configure myself.


I have tested CUCM ping the SP sip trunk IP.


I'd created a SIP trunk to the SP trunk sip IP in CUCM.


I create a new security profile and check "digest credentials"


I created a SIP realm with the next information.


realm: ip of the SP trunk sip. ( I dont have DNS configured in CUCM and I use the IP here, it is rght?)

User: user SP provided me.

Password: password SP provide me.


So I don' t know if I need enything else


It is not working anyway,


Thank you.


Best Regards

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Waldo Roca Martin Tue, 10/08/2013 - 07:20
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This is my notify


INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/TCP 10.10.76.253:50974;branch=z9hG4bK4bdfd210

From: "Gabriel Baron Losada" ;tag=0c27243078ac122d18e3cf27-00d00fb0

To:

Call-ID: [email protected]

Max-Forwards: 70

Date: Tue, 08 Oct 2013 13:44:48 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP8961/9.3.2

Contact:

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO

Remote-Party-ID: "Gabriel Baron Losada" ;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.0,X-cisco-xsi-8.0.1

Allow-Events: kpml,dialog

Content-Length: 404

Content-Type: application/sdp

Content-Disposition: session;handling=optional


v=0

o=Cisco-SIPUA 8180 0 IN IP4 10.10.76.253

s=SIP Call

t=0 0

m=audio 22242 RTP/AVP 0 8 18 102 9 116 124 101

c=IN IP4 10.10.76.253

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:102 L16/16000

a=rtpmap:9 G722/8000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv



This is the message the SP send me.


[51918,NET]
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 161.111.230.16:5060;branch=z9hG4bK23152c938da;received=161.111.230.16;rport=5060
From: ;tag=17176~425d8388-333a-4db7-8ae6-4c6137789d8d-36687858
To: ;tag=as679f84d4
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3cd91841"
Content-Length: 0

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